=== release 1.1.4 === 2013-08-28 Sebastian Dröge * configure.ac: releasing 1.1.4 2013-08-28 12:32:10 +0200 Sebastian Dröge * po/pt_BR.po: po: update translations 2013-08-27 15:25:16 +0200 Wim Taymans * gst/matroska/matroska-mux.c: matroska-mux: remove framerate restriction Remove the framerate restriction on the caps. 2013-08-27 09:38:16 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: only update next check time when reconsidering Don't update the next RTCP check time in all cases but only when we reconsidered. This avoids delaying sending a full RTCP packet when we are doing early feedback. 2013-08-27 09:37:33 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: add more debug 2013-08-27 09:34:46 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpsession.c: jitterbuffer: fix types of the retransmission event 2013-08-27 09:33:03 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: only timeout EXPECTED timers on gap Only timeout the EXPECTED timers when we detect a large seqnum gap. 2013-08-26 13:47:53 +0200 Sebastian Dröge * configure.ac: configure.ac: Don't set BZ2_LIBS if bz2 is not found 2013-08-26 11:50:27 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtsession: fix locking We need to take the session lock when getting and manipulating the source. 2013-08-26 11:50:13 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: rtpsession: add some more debug 2013-08-20 22:12:03 +0200 Mathieu Duponchelle * gst/videomixer/videomixer2.c: videomixer: don't send flush_stop twice. If we get flush start and a seek we need to only send flush_stop once. More info at #706441 2013-08-23 15:56:43 +0100 Tim-Philipp Müller * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: multipartdemux: propagate discont 2013-08-23 15:49:47 +0100 Tim-Philipp Müller * gst/multipart/multipartdemux.c: multipartdemux: remove dynamic sourcpads when going from PAUSED to READY 2013-08-23 15:29:28 +0100 Tim-Philipp Müller * gst/multipart/multipartdemux.c: * gst/multipart/multipartdemux.h: multipartdemux: timestamp output buffers based on first input buffer that provided bytes not last https://bugzilla.gnome.org/show_bug.cgi?id=637754 2013-08-23 15:47:25 +0200 Wim Taymans * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxqueue.h: rtxqueue: add property to configure queue size 2013-08-23 12:07:55 +0200 Wim Taymans * tests/examples/rtp/client-H264-rtx.sh: * tests/examples/rtp/server-VTS-H264-rtx.sh: tests: add retransmission example 2013-08-23 11:55:02 +0200 Wim Taymans * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpbin.h: rtpbin: proxy jitterbuffer do-retransmission property 2013-08-23 11:17:45 +0200 Michael Olbrich * gst/avi/gstavimux.c: avimux: unmap the correct buffer The audio buffer was mapped so unmap it and not the video buffer https://bugzilla.gnome.org/show_bug.cgi?id=706642 2013-08-18 23:32:22 -0400 Olivier Crête * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: pulsesink: Add property to find out the device currently in use https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-18 23:31:15 -0400 Olivier Crête * ext/pulse/pulsesink.c: pulsesink: De-duplicate code to get the current sink input info https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-18 22:27:37 -0400 Olivier Crête * ext/pulse/pulsesink.c: pulsesink: Implement changing the device while playing https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-18 23:32:22 -0400 Olivier Crête * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: pulsesrc: Add property to find out the device currently in use https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-18 23:31:15 -0400 Olivier Crête * ext/pulse/pulsesrc.c: pulsesrc: De-duplicate code to get the current source output info https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-18 22:27:37 -0400 Olivier Crête * ext/pulse/pulsesrc.c: pulsesrc: Implement changing the device while playing https://bugzilla.gnome.org/show_bug.cgi?id=590768 2013-08-22 14:55:14 +0200 Sebastian Dröge * configure.ac: configure: Fix bz2 configure check for Windows Due to function decorations on Windows AC_CHECK_LIB can't be used to check for bz2. https://bugzilla.gnome.org/show_bug.cgi?id=465924 2013-02-22 20:57:00 +0900 Akihiro Tsukada * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulseutil.c: * ext/pulse/pulseutil.h: pulsesink: Add support for AAC pass-through https://bugzilla.gnome.org/show_bug.cgi?id=694445 2013-06-24 17:29:37 +0200 Kishore Arepalli * ext/gdk_pixbuf/gstgdkpixbufoverlay.c: gdkpixbufoverlay: crashes if any property changes during playback when location property is not set https://bugzilla.gnome.org/show_bug.cgi?id=702988 2013-08-21 14:54:26 -0400 Olivier Crête * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: * ext/pulse/pulseutil.h: pulse: Share static caps definition between src and sink The src was also missing 24-bit sample formats 2013-08-21 16:53:59 +0200 Wim Taymans * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxqueue.h: rtx: various improvements Use locking Don't push from the event handler, collected packets in a queue and push from the chain function. Clear queues on shutdown. 2013-08-21 16:50:59 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: session: generate events correctly Do correct shifting of the bitmask for lost packets. 2013-08-21 16:47:40 +0200 Wim Taymans * gst/rtpmanager/gstrtpmanager.c: rtp: register rtx element better 2013-08-21 16:32:50 +0200 Sebastian Dröge * sys/directsound/gstdirectsoundsink.c: directsoundsink: WAVEFORMATEX is unsigned for 8 bit integers, and signed for others Probably fixes https://bugzilla.gnome.org/show_bug.cgi?id=705477 2013-08-21 13:03:34 +0100 Tim-Philipp Müller * ext/jpeg/gstjpegenc.c: jpegenc: don't ignore return value from _finish_frame() gst_video_encoder_finish_frame() will return FLOW_OK here if there's no output buffer. 2013-08-21 12:56:35 +0200 Wim Taymans * gst/rtp/gstrtpjpegdepay.c: jpegdepay: add some more debug 2013-08-21 12:10:00 +0200 Wim Taymans * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstdepay.h: rtpgstdepay: only push events when they changed Keep track of the STREAM_START and TAG events and only push them when they changed. 2013-08-21 10:52:59 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.c: rtpgstpay: taglists should not be merged in 1.0 2013-08-21 10:28:50 +0200 Wim Taymans * gst/rtp/gstrtpgstdepay.c: rtpgstdepay: flush on FLUSH_STOP event 2013-08-21 10:03:52 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.c: rtpgstpay: reset on state change Do full reset on state change to READY 2013-08-21 09:55:20 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.c: rtpgstpay: reset on FLUSH_STOP Clear the adapter and pending buffer list on FLUSH_STOP. 2013-08-21 09:39:30 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.c: rtpgstpay: don't use clock for config interval We can't use the clock to time our config-interval because we are not live (or there might not be a clock or the clock might not be running). Instead just simply take the timestamp diff. 2013-08-21 09:33:04 +0200 Wim Taymans * gst/rtp/gstrtpgstpay.h: rtpgstay: don't use // comments 2013-08-08 11:55:22 -0400 Youness Alaoui * gst/rtsp/gstrtspsrc.c: rtspsrc: Fix response argument in handle-request signal 2013-08-08 11:54:41 -0400 Youness Alaoui * gst/rtsp/gstrtspsrc.c: * gst/rtsp/gstrtspsrc.h: rtspsrc: Add sdes property and proxy it to rtpbin 2013-08-07 09:47:35 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: Send a stream-start whenever we send tags This is to make sure tags are cleared on the client if the stream-start was previously lost, otherwise, the client may end up with a merged taglist of multiple songs 2013-07-25 21:12:05 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: rtpgstpay: Add a config-interval property to resend the caps/tags at a regular interval This is useful in case the packet containing the inlined caps was lost or if new client joins an already running RTP stream and they missed the previous tag events. This also makes the payloader keep a list of merged tags so the retransmitted tag event contains all previously received. A STREAM_START event will flush the list of tags. 2013-07-25 21:10:10 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time 2013-07-25 21:03:34 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps 2013-07-25 20:54:50 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: * gst/rtp/gstrtpgstpay.h: rtpgstpay: Add a create_from_adapter API and use a list of GstBufferList This is necessary to fix event/caps sending. If we send a STREAM_START packet, it will cause an error because the stream didn't receive its caps and new-segment events, so we must wait for the first buffer before sending the stream-start event buffer. However, the caps will be sent at the same time and so the 'inline caps' will be set for the event. We need to be able to payload individual packets (data, caps or events) and only send them when we call flush. 2013-07-25 17:56:38 -0400 Youness Alaoui * gst/rtp/gstrtpgstdepay.c: * gst/rtp/gstrtpgstpay.c: rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START 2013-07-25 17:52:16 -0400 Youness Alaoui * gst/rtp/gstrtpgstpay.c: rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3 2013-08-20 14:36:59 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: handle EOS When the queue is empty, and we received EOS, pause and push an EOS event downstream. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706387 2013-08-20 10:26:15 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: update docs 2013-08-20 10:25:17 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: update all timers Keep looping over all registered timers so that we can mark them lost instead of stopping as soon as we find the timer for the current seqnum. 2013-08-20 08:55:50 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: remove unused variables 2013-08-19 21:10:00 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: reorganize timer handling Restructure handling of incomming packet and the gap with the expected seqnum and register all timers from the _chain function. Convert a timer to a LOST packet timer when the max amount of retransmission requests has been reached. 2013-08-19 21:37:44 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: refactor packet spacing calculation 2013-08-19 21:34:38 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: keep track of last seqnum and dts 2013-08-19 21:29:49 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: small cleanups 2013-08-19 21:21:08 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: reset retransmission timers in add/reschedule Reset the retransmission timers when adding and rescheduling a timer. 2013-08-19 21:12:13 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: rename variables for packet spacing 2013-08-19 14:58:01 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: remove lost timer when we get the packet When we receive a packet, also remove the LOST timer for it. 2013-08-19 14:56:49 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: expected seqnum must increase Only update the expected seqnum when it is bigger than the previous expected seqnum. 2013-08-19 14:55:49 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: add more debug 2013-08-12 16:15:54 +0200 Wim Taymans * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpmanager.c: * gst/rtpmanager/gstrtprtxqueue.c: * gst/rtpmanager/gstrtprtxqueue.h: rtxqueue: add retransmission queue element 2013-08-12 14:53:33 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: add some docs 2013-08-06 16:29:54 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: handle NACK feedback and generate events Handle and parse the feedback NACK packets and generate a Retransmission event for each NACKed packet 2013-08-19 13:19:42 -0400 Olivier Crête * sys/v4l2/gstv4l2object.c: v4l2: Add forward declaration for gst_v4l2_object_get_format_list 2012-10-22 17:58:07 -0400 Olivier Crête * sys/v4l2/gstv4l2object.c: * sys/v4l2/gstv4l2object.h: * sys/v4l2/gstv4l2sink.c: * sys/v4l2/gstv4l2sink.h: * sys/v4l2/gstv4l2src.c: * sys/v4l2/gstv4l2src.h: v4l2: De-duplicate caps probing between src and sink 2013-08-13 17:32:17 -0400 Olivier Crête * ext/pulse/Makefile.am: * ext/pulse/pulseprobe.c: * ext/pulse/pulseprobe.h: * ext/pulse/pulsesink.c: * ext/pulse/pulsesink.h: * ext/pulse/pulsesrc.c: * ext/pulse/pulsesrc.h: pulse: Remove unused GstPulseProbe 2013-08-19 12:46:45 -0400 Olivier Crête * sys/v4l2/gstv4l2tuner.c: * sys/v4l2/tuner.c: * sys/v4l2/tunerchannel.c: * sys/v4l2/tunernorm.c: v4l2: Use G_DEFINE_ macros for added thread safety 2013-08-17 11:28:13 +0200 Thibault Saunier * gst/videomixer/videomixer2.c: * gst/videomixer/videomixer2.h: videomixer: Do not send flush_stop ourself after a flush_start When we receive a flush_start, we should wait for the next flush_stop and foward it, not create a flush_stop ourself. 2013-08-16 17:10:31 +0200 Wim Taymans * gst/rtp/gstrtph264depay.c: h264depay: init debug category early Init the debug variable when we register the element because it is also used by the payloader element when it calls the add_sps_pps method. 2013-08-16 13:26:28 +0200 Sebastian Dröge * ext/flac/gstflacenc.c: flacenc: Properly set headers via the base class instead of just pushing them downstream Prevents buffers from being send before the caps and segment events. 2013-08-15 10:59:10 +0100 Chris Bass * gst/isomp4/qtdemux.c: qtdemux: check denominator isn't zero before scaling duration. When gst_qtdemux_configure_stream sets fps_d, check that n_samples is non-zero before using it as a denominator to scale the stream duration. https://bugzilla.gnome.org/show_bug.cgi?id=706076 2013-08-15 15:08:05 +0200 Sebastian Dröge * ext/jpeg/gstjpegdec.c: * ext/jpeg/gstjpegenc.c: * ext/libpng/gstpngdec.c: * ext/vpx/gstvp8dec.c: * ext/vpx/gstvp9dec.c: ext: Use new flush vfunc of video codec base classes and remove reset implementations 2013-08-14 16:19:32 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: forward flush before stopping dataflow First forward the flush event and then stop our loop function. 2013-08-14 13:10:32 +0100 Tim-Philipp Müller * configure.ac: configure: require libsoup >= 2.38 Bump libsoup requirement for newer API used, like headers_get_one(). 2.38 is from early 2012 and is in linen with our GLib requirement. 2013-08-14 11:54:19 +0100 Tim-Philipp Müller * ext/soup/gstsouphttpsrc.c: soup: don't use deprecated soup_message_headers_get() API 2013-08-13 17:44:50 +0200 Edward Hervey * .gitignore: .gitignore: Ignore files from automake test-driver 2013-08-12 15:28:34 -0400 Olivier Crête * gst/rtp/gstrtph264pay.c: * gst/rtp/gstrtph264pay.h: rtph264pay: Use the SPS/PPS handling function from the depayloader Remove duplicated copies https://bugzilla.gnome.org/show_bug.cgi?id=705553 2013-08-12 15:26:08 -0400 Olivier Crête * gst/rtp/gstrtph264depay.c: * gst/rtp/gstrtph264depay.h: rtph264depay: Make the SPS/PPS deduplication function generic Make it not touch any internals of the depayloader https://bugzilla.gnome.org/show_bug.cgi?id=705553 2013-08-13 14:09:20 +0100 Chris Bass * gst/audioparsers/gstaacparse.c: aacparse: allow conversion from raw AAC to ADTS This patch will prepend ADTS headers to raw AAC audio frames, allowing upstream elements to link to decoders that only support AAC in ADTS format. Note that no error correction bits are added to ADTS frames in this code. https://bugzilla.gnome.org/show_bug.cgi?id=615740 2013-08-13 12:44:11 +0200 Sebastian Dröge * gst/rtsp/gstrtspsrc.c: rtspsrc: Only free GCheckSum after its last usage https://bugzilla.gnome.org/show_bug.cgi?id=705760 2013-08-13 12:02:29 +0200 Andoni Morales Alastruey * ext/soup/gstsouphttpsrc.c: souphttpsrc: fix critical setting a NULL uri redirection 2013-07-13 01:50:56 +0200 Andoni Morales Alastruey * ext/soup/gstsouphttpsrc.c: * ext/soup/gstsouphttpsrc.h: souphttpsrc: add redirection to the URI query 2013-07-31 10:42:07 +0200 Matej Knopp * gst/isomp4/qtdemux.c: qtdemux: elst should offset samples instead of buffers The current approach where buffers are offset is not ideal, as during seek and loop current time is compared to sample times. https://bugzilla.gnome.org/show_bug.cgi?id=700264 2013-08-07 19:32:07 +0200 Thibault Saunier * gst/videomixer/videomixer2.c: * tests/check/elements/videomixer.c: videomixer: Send EOS if buf_end >= segment.stop That means the whole segment is already played, and we are sure we are EOS at that point. Also handle segment seeks, and do not send EOS in that case. 2013-08-04 14:40:38 +0200 Matej Knopp * gst/avi/gstavidemux.c: avidemux: send proper stream_start event https://bugzilla.gnome.org//show_bug.cgi?id=705449 2013-08-08 11:51:17 +0200 Sebastian Dröge * gst/matroska/ebml-read.c: * gst/matroska/matroska-demux.c: matroskademux: Don't print warnings during flushing and stop as soon as possible https://bugzilla.gnome.org//show_bug.cgi?id=705442 2013-08-07 11:14:38 +0100 Tim-Philipp Müller * gst/rtp/gstrtpvp8depay.c: rtpvp8depay: mark key frames and delta frames properly https://bugzilla.gnome.org/show_bug.cgi?id=705550 2013-08-05 23:23:57 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: add NACK feedback in RTCP 2013-08-05 23:22:16 +0200 Wim Taymans * gst/rtpmanager/rtpsource.c: * gst/rtpmanager/rtpsource.h: source: add methods to register NACK Add a method to register a missing packet for an ssrc along with methods to get the missing packets and clear them. 2013-08-04 23:05:36 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: handle Retransmission event and schedule NACK Handle the retransmission event from downstream and use it to schedule a NACK request. 2013-08-05 23:20:29 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: pass data to remove func Pass the data to the remove function because we are going to deref it when there is pli or fir. 2013-08-06 15:28:50 +0200 Thibault Saunier * gst/isomp4/qtdemux.c: qtdemux: Fix compilation 2013-08-06 15:17:44 +0200 Thibault Saunier * gst/isomp4/qtdemux.c: qtdemux: Raw buffer DTS should always be CLOCK_TIME_NONE 2013-08-06 11:58:38 +0200 Thibault Saunier * gst/videomixer/videomixer2.c: videomixer: Make sure to send EOS if the buffer end time equals the segment end time Otherwize EOS never gets sent in that particular case. 2013-08-05 08:49:50 +0200 Sjoerd Simons * gst/goom/gstgoom.c: goom: Ensure src caps are writable In some cases the src caps determined by goom weren't writable, causing a bunch of assertion failures and failed caps. Fixed by always explicitely making the caps writable https://bugzilla.gnome.org/show_bug.cgi?id=705475 2013-08-04 23:18:29 +0200 Wim Taymans * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: use common send_rtcp method Reuse the send_rtcp method that already asks for the current time when requesting a keyframe. 2013-08-04 23:12:50 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: * gst/rtpmanager/rtpsession.h: session: Don't use ClockTimeDiff for unsigned delays 2013-08-04 16:52:15 +0200 Edward Hervey * gst/isomp4/gstqtmux.c: qtmux: Use buffer PTS if DTS is not set Avoids ending up with completely bogus scaled duration/pts when new buffers have invalid DTS. 2013-08-04 14:32:47 +0100 Tim-Philipp Müller * tests/check/elements/souphttpsrc.c: tests: skip https test if there's no TLS support in soup/glib 2013-08-04 11:20:41 +0100 Tim-Philipp Müller * gst/rtsp/gstrtpdec.c: rtpdec: use generic marshaller 2013-08-04 10:52:33 +0100 Tim-Philipp Müller * Makefile.am: * sys/v4l2/.gitignore: * sys/v4l2/Makefile.am: * sys/v4l2/gstv4l2-marshal.list: * sys/v4l2/tuner-marshal.list: * sys/v4l2/tuner.c: * sys/v4l2/tuner.h: * win32/MANIFEST: * win32/common/tuner-enumtypes.c: * win32/common/tuner-enumtypes.h: * win32/common/tuner-marshal.c: * win32/common/tuner-marshal.h: v4l2: remove unused enumtypes and use generic marshaller 2013-08-04 10:47:38 +0100 Tim-Philipp Müller * Makefile.am: * gst/udp/.gitignore: * win32/common/gstudp-enumtypes.c: * win32/common/gstudp-enumtypes.h: * win32/common/gstudp-marshal.c: * win32/common/gstudp-marshal.h: udp: remove unused marshal and enumtypes files 2013-08-04 09:38:19 +0100 Tim-Philipp Müller * Makefile.am: * gst/rtpmanager/.gitignore: * gst/rtpmanager/Makefile.am: * gst/rtpmanager/gstrtpbin-marshal.list: * gst/rtpmanager/gstrtpbin.c: * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/gstrtpptdemux.c: * gst/rtpmanager/gstrtpsession.c: * gst/rtpmanager/gstrtpssrcdemux.c: * gst/rtpmanager/rtpsession.c: * win32/MANIFEST: * win32/common/gstrtpbin-marshal.c: * win32/common/gstrtpbin-marshal.h: rtpmanager: use generic marshaller 2013-08-04 00:13:07 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: send event in right direction 2013-08-02 17:38:34 -0700 David Schleef * configure.ac: * tests/check/Makefile.am: tests: create/remove orc directory at proper time Before automake creates .deps directories, and during distclean. 2013-08-03 00:25:44 +0200 Wim Taymans * gst/rtpmanager/rtpsession.c: session: add FIR and PLI like other RTCP packets Add the FIR and PLI packets like the other RTCP packet instead of from the on-sending-rtcp default signal handler. 2013-08-02 17:22:55 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: fix property ranges 2013-08-02 16:42:52 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: push retransmission events 2013-08-02 14:12:16 +0200 Lubosz Sarnecki * configure.ac: build: add subdir-objects to AM_INIT_AUTOMAKE Fixes warnings with automake 1.14 https://bugzilla.gnome.org/show_bug.cgi?id=705350 2013-08-02 14:54:56 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: add support for retransmission retry When we didn't receive a packet after requesting retransmission, retry asking for retransmission for a certain period. 2013-08-02 14:19:54 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: add properties Add properties to control retransmission parameters 2013-08-02 12:44:58 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: use corrected timeout when rescheduling When we recalculate the timeout, use the corrected timeout value depending on the timer type. 2013-08-02 12:43:00 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: update timers after queueing Else we might update the timer needlessly for duplicates. 2013-08-02 12:42:08 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: move method up 2013-08-02 06:28:32 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: small cleanup 2013-08-01 23:26:06 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: unschedule old expected packets When we receive a new packet, unschedule old outstanding packets when their seqnum is too far away. 2013-08-01 23:29:23 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: refactor timer update 2013-08-01 23:24:29 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: update timers when removing Update the timers when we remove a timer. Handle canceled timers, make them unschedule the current timer and trigger the timeout code. 2013-08-01 23:22:02 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: fix typo 2013-08-01 15:40:52 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: improve timeout management If we change the seqnum of an existing timer and we were waiting for that timer, unschedule it. If we change the timeout of an existing timer and we were waiting on it, only unschedule when the new time is smaller. 2013-08-01 15:05:35 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: install timer for expected arrival Install a timer that is triggered when the expected arrival time of a packet expired. 2013-08-01 14:56:00 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: improve unschedule of timers Conflicts: gst/rtpmanager/gstrtpjitterbuffer.c 2013-08-01 12:21:53 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: move code around 2013-08-01 12:07:11 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: estimate inter packet spacing When we see two packets with consecutive seqnums and a different RTP time, use the DTS difference as the inter packet spacing estimate. 2013-08-01 12:01:15 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: keep track of current timeout 2013-08-01 11:49:10 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: cleanup timer handling 2013-08-01 11:40:41 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: reset is only possible with a GAP 2013-08-01 11:29:32 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: * gst/rtpmanager/rtpjitterbuffer.c: jitterbuffer: operate on DTS Make the jitterbuffer schedule the timeouts based on the DTS instead of the PTS. This makes it all smoother with reordered frames and gives the decoder time to reorder the frames in time. 2013-08-01 11:14:12 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: rename timout variable 2013-07-31 17:08:58 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: small cleanup 2013-07-31 16:59:58 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: block output in paused or buffering 2013-07-31 16:59:09 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: store pts in timer Only store the pts in the timer so that we can both do timeouts with timings on the input and output of the jitterbuffer. 2013-07-30 23:14:24 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: rtpjitterbuffer: refactor jitterbuffer Refactor the jitterbuffer code. Make separate function for peeking a buffer, pushing the next buffer, waiting for timeouts and handling the timeouts. The main loop now tries to push as many buffers as it can until it runs out of buffers or when it detects a seqnum discont. Then it will wait for some event to happen before attempting to push more buffers. Make methods to register timeouts in an array. These timeouts are registered when we detect a missing packet, sync for the first packet or when we find an estimation for the end-of-stream. This greatly simplifies and clarifies the code and also makes it possible to register more complicated timeout schemes later. 2013-07-30 18:52:58 +0200 Wim Taymans * gst/rtpmanager/rtpjitterbuffer.c: rtpjitterbuffer: use NULL to ignore percent If we pass NULL to pop and push we ignore the percent result. 2013-07-30 07:00:19 +0200 Wim Taymans * gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer: refactor Move eos estimation into separate function 2013-07-30 14:28:19 +0100 Tim-Philipp Müller * gst/flv/gstflvdemux.c: flvdemux: don't leak stream_id string https://bugzilla.gnome.org/show_bug.cgi?id=705142 2013-07-29 19:53:52 +0100 Tim-Philipp Müller * po/LINGUAS: * po/da.po: * po/de.po: * po/el.po: * po/gl.po: * po/hr.po: * po/hu.po: * po/ja.po: * po/nb.po: * po/nl.po: * po/pl.po: * po/ru.po: * po/sl.po: * po/tr.po: * po/uk.po: * po/vi.po: * po/zh_CN.po: po: update translations 2013-07-29 19:48:54 +0100 Tim-Philipp Müller * tests/check/elements/.gitignore: tests: ignore new test binaries 2013-07-29 14:47:49 +0200 Sebastian Dröge * configure.ac: Back to development